1. Field of the Invention
The present invention relates to a voice transceiver for use in voice transmission by means of digital voice signals which utilize compressed voice coding.
This application is based on Patent Application No. Hei 10-144734 filed in Japan.
2. Relevant Art
FIG. 3 is a block diagram showing the structure of a conventional voice transceiver. In the figure, a decoding code buffer 301 receives compressed voice codes from a circuit (not shown in the figure), and stores these codes into an internal memory. A voice decoder 401 then digitalizes and expands these compressed voice codes stored in the memory of the decoding code buffer 301 into digital voice data.
An SP (speaker) output buffer 501 inputs and stores the voice data expanded by means of the voice decoder 401. D/A converter 601 converts the digital voice data stored in the SP output buffer 501 into an analog voice signal. An amplifier 701 amplifies the analog signal by a predetermined magnitude (amplification), and a speaker 801 then emits the amplified analog voice signal into the air.
In addition, a microphone 802 (hereinafter also referred to as “mic”) collects a transmitted voice and converts it into an electronic signal. The microphone 802 then converts the aforementioned conversion result into an analog voice input signal. An amplifier 702 amplifies the analog voice input signal to be inputted by a predetermined magnitude (amplification). A/D converter 602 then converts the analog voice input signal into a digital voice input signal. MIC (microphone) input buffer 502 subsequently stores this digital voice input signal.
A voice encoder 402 encodes the digital voice input signal stored in the MIC input buffer 502, and outputs a compressed voice code as a result of the encoding. A compression code buffer 302 then stores the compressed voice code inputted from the voice encoder 402.
In the following, an operation of the voice transceiver according to aforementioned conventional example will be described.
For example, the decoding code buffer 301 temporarily stores the compressed voice code inputted from a communication circuit (not shown in the figures) into an internal memory portion. Subsequently, using the storage of the compressed voice code into decoding code buffer 301 as a trigger, the voice decoder 401 begins processing by expanding the compressed voice code stored in the memory portion of the decoding code buffer 301, and generating digitalized digital voice data.
In this manner, the generated voice data is inputted and written into the SP output buffer 501. On the other hand, a voice encoder 402 detects the writing of the digital voice data required for encoding one frame in MIC input buffer 502. The voice encoder 402 then commences operation by compressing the digital voice data, and generating a compressed voice code.
Following completion of this operation, the voice encoder 402 outputs the generated compressed voice code to a compression code buffer 302. This compression code buffer 302 then stores the inputted compressed voice code. In this manner, the compression code buffer 302 transmits the compressed voice code stored therein to the communication circuit side (not shown in the figures).
Furthermore, the right side of the operation from the D/A converter 601 and A/D converter 602, respectively shown in FIG. 3, is performed during a fixed clock cycle by means of hardware. In other words, the digital voice data of SP output buffer 501 is outputted one sample at a time when necessary, and converted into an analog voice signal by means of D/A converter 601. In addition, at the same time, the analog voice signal inputted from the microphone 802 is sampled when necessary during a fixed cycle, converted by A/D converter 602 into a voice signal, and written into MIC input buffer 502 as necessary.
However, in the case when the aforementioned voice transceiver is operated in an environment in which the digital voice input signal, which serves as the reception decoding code, is not smoothly and regularly supplied, problems arise such as the generation of interruptions in the output voice from speaker 801, leading to an extreme degradation of the quality of the voice reception (receiving voice quality).
For example, in a personal computer, the supply of a smooth and regular reception signal code cannot be guaranteed by controlling the processing assignment of the processor, by means of processing of the aforementioned voice transceiver (using the same processor) and simultaneously operating an optional user software. As a result, as described above, extreme degradation of the voice reception quality results, when embodying the processing of the aforementioned voice transceiver as software of a personal computer, using a voice transmitter, desktop conference system or the like, which utilizes a personal computer.
In addition, in a multi-media transmission terminal, besides voice codes, other data such as images and the like are mixed therein and transmitted. As a result, when the transmission data in a communication circuit is disrupted, it is not possible to specify the disrupted data as voice data or otherwise, and thus a regular and unimpaired supply of reception decoding codes cannot be guaranteed. Consequently, the quality of voice reception is notably degraded even in the voice transmission processing portion of a multi-media transmission terminal.
Here, enlargement of the SP output buffer and absorption of the jitter from the output voice may be considered, as an example of a method for avoiding the degradation of the voice reception quality occurring in a voice transmission processing portion of a multimedia transmission terminal. However, an increase in the SP output buffer causes an increase in the shift distance over which the digital voice input signal must pass from the point of input to the point of output. This aspect, in turn, leads to a delay in the voice, and is hence undesirable from a practical standpoint.
In addition, the jitter amount is statistically distributed. As a result, there is a distinct disadvantage in that it is not possible to calculate an absolute value with respect to the optimal amount for enlarging the SP output buffer, as this value changes depending on various conditions.
Consequently, as a result of the reception voice signal not being supplied after monitoring the remaining data of the SP output buffer, the conventional technology poses problems in that in an environment in which the supply of the reception decoding code is performed in a “burst transmission” manner, when the supply of the receiving decoding code is interrupted, or alternatively when the supply of the receiving decoding code is continued after such an interruption, the SP output voice is similarly interrupted and non-continuous, thereby leading to extreme degradation of the voice reception quality.